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SipToSis TM - SIP to Skype Gateway Bridge Proxy Adapter Converter Interconnect
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What is the SipToSis Skype Gateway?
SipToSis (Sip to Skype integration software) is Java software that allows you to make and receive Skype calls from your SIP/VOIP telephone adapter, SIP IP phone or Asterisk/SIP PBX. You can also make and receive SIP calls from Skype. It let's you integrate Skype into your SIP VOIP phone system. Basically a Skype/SIP Bridge/Gateway/Proxy/Adapter/Converter. It has a codec converter to convert SIP RTP audio to compatible Skype PCM audio and Skype PCM audio to SIP RTP audio. It performs SIP signaling and Skype call handling to connect with your SIP adapter, Asterisk Server, SIP PBX or SIP VOIP provider.

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SipToSis SIP to Skype Gateway Features:
  • Call Skype users using mappings/speed dial or use SkypeOut to make PSTN calls from a SIP device
  • Make SIP calls from another Skype user using a SIP provider or SIP PBX (SkypeCaller-->Skype/SipToSis Host-->SIP)
  • Skype callers can be directed to the SIP address of your choice
  • SIP callers can be directed to the Skype user of your choice
  • SIP to Skype authentication/denial mappings via SIP caller ID and IP blocks
  • Skype to SIP authentication/denial mappings via incoming Skype User ID
  • Automatic authorization of new contacts as of version 20110324 - feature must be enabled
  • SIP DTMF touchtone decoding/encoding via RFC2833, INFO or Inband
  • Skype DTMF touchtone decoding/encoding via Inband
  • Connect Asterisk, FreePBX, Elastix, trixbox, PBX-in-a-Flash, 3CX or other SIP PBX to Skype Users
  • Conference call as of version 20091115 - See FAQ page for compatibility
  • Callback capability as of version 20091115 - See FAQ page for compatibility
  • Skype voicemail retrieval via SIP as of version 20091206
  • Outgoing Skype voicemail support as of version 20091206 - See FAQ page for compatibility
  • Auto play pre-recorded file(s) to SIP and Skype callers
  • SIP caller pin authentication and dialing
  • Skype caller pin authentication and dialing
  • PCMU (u-law)/PCMA (a-law)/G.711/iLBC built in codecs and GSM with additional libs
  • L16/16000 built in codec as of version 20110421
  • Codec interface so you can add other codecs
  • SIP and Skype Hold
  • SkypeOut dialing rules - customize for your location
  • SIP outbound dialing rules - customize for your location
  • STUN support for public IP discovery
  • Call Time limiting ability
  • Usage limiting abilities by used time and unique called number count
  • Can be setup as a multi channel Skype to Asterisk Trunk for multiple simultaneous calls with the stsTrunkBuilder
  • Multiplatform (Windows/Linux/Mac OS X)
  • Run everything on a single computer if you wish
  • Windows users can run it as a service - see Appicus Windows Service Wrapper


SipToSis SIP to Skype Gateway System Requirements:
  • Skype client - See FAQ page for known working versions
  • Sun/Oracle's Java 1.5 or higher (Linux users - Do NOT use openjdk)
  • SIP/VOIP adapter such as a SPA3102, SIP IP Phone, register with a SIP provider or setup an Asterisk/SIP PBX server
  • Skype4Java compatible platform
  • Sufficient bandwidth to support your configuration - Broadband preferred.
  • Skype compatible sound device
This product uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype.

SipToSis is GNU GPL Licensed software.

Online documentation:
NOTE: Any support will be on the SipToSis Forum. Please look at Common problems or the SipToSis FAQ to see if your problem/question hasn't already been answered. You can keep up with the latest by subscribing to the forum RSS feed.